gstrtppcmudepay.c \
gstrtppcmupay.c \
gstrtppcmapay.c \
+ gstrtpg722depay.c \
+ gstrtpg722pay.c \
gstrtpg723depay.c \
gstrtpg723pay.c \
gstrtpg726pay.c \
gstrtppcmudepay.h \
gstrtppcmupay.h \
gstrtppcmapay.h \
+ gstrtpg722depay.h \
+ gstrtpg722pay.h \
gstrtpg723depay.h \
gstrtpg723pay.h \
gstrtpg726depay.h \
#include "gstrtppcmapay.h"
#include "gstrtppcmadepay.h"
#include "gstrtppcmudepay.h"
+#include "gstrtpg722depay.h"
+#include "gstrtpg722pay.h"
#include "gstrtpg723depay.h"
#include "gstrtpg723pay.h"
#include "gstrtpg726depay.h"
if (!gst_rtp_ilbc_depay_plugin_init (plugin))
return FALSE;
+ if (!gst_rtp_g722_depay_plugin_init (plugin))
+ return FALSE;
+
+ if (!gst_rtp_g722_pay_plugin_init (plugin))
+ return FALSE;
+
if (!gst_rtp_g723_depay_plugin_init (plugin))
return FALSE;
--- /dev/null
+/* GStreamer
+ * Copyright (C) <2007> Wim Taymans <wim.taymans@gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include <string.h>
+#include <stdlib.h>
+
+#include <gst/audio/audio.h>
+#include <gst/audio/multichannel.h>
+
+#include "gstrtpg722depay.h"
+#include "gstrtpchannels.h"
+
+GST_DEBUG_CATEGORY_STATIC (rtpg722depay_debug);
+#define GST_CAT_DEFAULT (rtpg722depay_debug)
+
+static GstStaticPadTemplate gst_rtp_g722_depay_src_template =
+GST_STATIC_PAD_TEMPLATE ("src",
+ GST_PAD_SRC,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS ("audio/G722, "
+ "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]")
+ );
+
+static GstStaticPadTemplate gst_rtp_g722_depay_sink_template =
+ GST_STATIC_PAD_TEMPLATE ("sink",
+ GST_PAD_SINK,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS ("application/x-rtp, "
+ "media = (string) \"audio\", "
+ "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
+ "clock-rate = (int) 8000, "
+ /* "channels = (int) [1, MAX]" */
+ /* "channel-order = (string) ANY" */
+ "encoding-name = (string) \"G722\";"
+ "application/x-rtp, "
+ "media = (string) \"audio\", "
+ "payload = (int) " GST_RTP_PAYLOAD_G722_STRING ", "
+ "clock-rate = (int) [ 1, MAX ]"
+ /* "channels = (int) [1, MAX]" */
+ /* "emphasis = (string) ANY" */
+ /* "channel-order = (string) ANY" */
+ )
+ );
+
+GST_BOILERPLATE (GstRtpG722Depay, gst_rtp_g722_depay, GstBaseRTPDepayload,
+ GST_TYPE_BASE_RTP_DEPAYLOAD);
+
+static gboolean gst_rtp_g722_depay_setcaps (GstBaseRTPDepayload * depayload,
+ GstCaps * caps);
+static GstBuffer *gst_rtp_g722_depay_process (GstBaseRTPDepayload * depayload,
+ GstBuffer * buf);
+
+static void
+gst_rtp_g722_depay_base_init (gpointer klass)
+{
+ GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
+
+ gst_element_class_add_pad_template (element_class,
+ gst_static_pad_template_get (&gst_rtp_g722_depay_src_template));
+ gst_element_class_add_pad_template (element_class,
+ gst_static_pad_template_get (&gst_rtp_g722_depay_sink_template));
+
+ gst_element_class_set_details_simple (element_class, "RTP audio depayloader",
+ "Codec/Depayloader/Network",
+ "Extracts G722 audio from RTP packets",
+ "Wim Taymans <wim.taymans@gmail.com>");
+}
+
+static void
+gst_rtp_g722_depay_class_init (GstRtpG722DepayClass * klass)
+{
+ GstBaseRTPDepayloadClass *gstbasertpdepayload_class;
+
+ gstbasertpdepayload_class = (GstBaseRTPDepayloadClass *) klass;
+
+ gstbasertpdepayload_class->set_caps = gst_rtp_g722_depay_setcaps;
+ gstbasertpdepayload_class->process = gst_rtp_g722_depay_process;
+
+ GST_DEBUG_CATEGORY_INIT (rtpg722depay_debug, "rtpg722depay", 0,
+ "G722 RTP Depayloader");
+}
+
+static void
+gst_rtp_g722_depay_init (GstRtpG722Depay * rtpg722depay,
+ GstRtpG722DepayClass * klass)
+{
+ /* needed because of GST_BOILERPLATE */
+}
+
+static gint
+gst_rtp_g722_depay_parse_int (GstStructure * structure, const gchar * field,
+ gint def)
+{
+ const gchar *str;
+ gint res;
+
+ if ((str = gst_structure_get_string (structure, field)))
+ return atoi (str);
+
+ if (gst_structure_get_int (structure, field, &res))
+ return res;
+
+ return def;
+}
+
+static gboolean
+gst_rtp_g722_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps)
+{
+ GstStructure *structure;
+ GstRtpG722Depay *rtpg722depay;
+ gint clock_rate, payload, samplerate;
+ gint channels;
+ GstCaps *srccaps;
+ gboolean res;
+ const gchar *channel_order;
+ const GstRTPChannelOrder *order;
+
+ rtpg722depay = GST_RTP_G722_DEPAY (depayload);
+
+ structure = gst_caps_get_structure (caps, 0);
+
+ payload = 96;
+ gst_structure_get_int (structure, "payload", &payload);
+ switch (payload) {
+ case GST_RTP_PAYLOAD_G722:
+ channels = 1;
+ clock_rate = 8000;
+ samplerate = 16000;
+ break;
+ default:
+ /* no fixed mapping, we need clock-rate */
+ channels = 0;
+ clock_rate = 0;
+ samplerate = 0;
+ break;
+ }
+
+ /* caps can overwrite defaults */
+ clock_rate =
+ gst_rtp_g722_depay_parse_int (structure, "clock-rate", clock_rate);
+ if (clock_rate == 0)
+ goto no_clockrate;
+
+ if (clock_rate == 8000)
+ samplerate = 16000;
+
+ if (samplerate == 0)
+ samplerate = clock_rate;
+
+ channels =
+ gst_rtp_g722_depay_parse_int (structure, "encoding-params", channels);
+ if (channels == 0) {
+ channels = gst_rtp_g722_depay_parse_int (structure, "channels", channels);
+ if (channels == 0) {
+ /* channels defaults to 1 otherwise */
+ channels = 1;
+ }
+ }
+
+ depayload->clock_rate = clock_rate;
+ rtpg722depay->rate = samplerate;
+ rtpg722depay->channels = channels;
+
+ srccaps = gst_caps_new_simple ("audio/G722",
+ "rate", G_TYPE_INT, samplerate, "channels", G_TYPE_INT, channels, NULL);
+
+ /* add channel positions */
+ channel_order = gst_structure_get_string (structure, "channel-order");
+
+ order = gst_rtp_channels_get_by_order (channels, channel_order);
+ if (order) {
+ gst_audio_set_channel_positions (gst_caps_get_structure (srccaps, 0),
+ order->pos);
+ } else {
+ GstAudioChannelPosition *pos;
+
+ GST_ELEMENT_WARNING (rtpg722depay, STREAM, DECODE,
+ (NULL), ("Unknown channel order '%s' for %d channels",
+ GST_STR_NULL (channel_order), channels));
+ /* create default NONE layout */
+ pos = gst_rtp_channels_create_default (channels);
+ gst_audio_set_channel_positions (gst_caps_get_structure (srccaps, 0), pos);
+ g_free (pos);
+ }
+
+ res = gst_pad_set_caps (depayload->srcpad, srccaps);
+ gst_caps_unref (srccaps);
+
+ return res;
+
+ /* ERRORS */
+no_clockrate:
+ {
+ GST_ERROR_OBJECT (depayload, "no clock-rate specified");
+ return FALSE;
+ }
+}
+
+static GstBuffer *
+gst_rtp_g722_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
+{
+ GstRtpG722Depay *rtpg722depay;
+ GstBuffer *outbuf;
+ gint payload_len;
+ gboolean marker;
+
+ rtpg722depay = GST_RTP_G722_DEPAY (depayload);
+
+ payload_len = gst_rtp_buffer_get_payload_len (buf);
+
+ if (payload_len <= 0)
+ goto empty_packet;
+
+ GST_DEBUG_OBJECT (rtpg722depay, "got payload of %d bytes", payload_len);
+
+ outbuf = gst_rtp_buffer_get_payload_buffer (buf);
+ marker = gst_rtp_buffer_get_marker (buf);
+
+ if (marker) {
+ /* mark talk spurt with DISCONT */
+ GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
+ }
+
+ return outbuf;
+
+ /* ERRORS */
+empty_packet:
+ {
+ GST_ELEMENT_WARNING (rtpg722depay, STREAM, DECODE,
+ ("Empty Payload."), (NULL));
+ return NULL;
+ }
+}
+
+gboolean
+gst_rtp_g722_depay_plugin_init (GstPlugin * plugin)
+{
+ return gst_element_register (plugin, "rtpg722depay",
+ GST_RANK_MARGINAL, GST_TYPE_RTP_G722_DEPAY);
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) <2010> Wim Taymans <wim.taymans@gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+
+#ifndef __GST_RTP_G722_DEPAY_H__
+#define __GST_RTP_G722_DEPAY_H__
+
+#include <gst/gst.h>
+#include <gst/rtp/gstbasertpdepayload.h>
+
+G_BEGIN_DECLS
+
+/* Standard macros for defining types for this element. */
+#define GST_TYPE_RTP_G722_DEPAY \
+ (gst_rtp_g722_depay_get_type())
+#define GST_RTP_G722_DEPAY(obj) \
+ (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTP_G722_DEPAY,GstRtpG722Depay))
+#define GST_RTP_G722_DEPAY_CLASS(klass) \
+ (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTP_G722_DEPAY,GstRtpG722DepayClass))
+#define GST_IS_RTP_G722_DEPAY(obj) \
+ (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_G722_DEPAY))
+#define GST_IS_RTP_G722_DEPAY_CLASS(klass) \
+ (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_G722_DEPAY))
+
+typedef struct _GstRtpG722Depay GstRtpG722Depay;
+typedef struct _GstRtpG722DepayClass GstRtpG722DepayClass;
+
+/* Definition of structure storing data for this element. */
+struct _GstRtpG722Depay
+{
+ GstBaseRTPDepayload depayload;
+
+ guint rate;
+ guint channels;
+};
+
+/* Standard definition defining a class for this element. */
+struct _GstRtpG722DepayClass
+{
+ GstBaseRTPDepayloadClass parent_class;
+};
+
+GType gst_rtp_g722_depay_get_type (void);
+
+gboolean gst_rtp_g722_depay_plugin_init (GstPlugin * plugin);
+
+G_END_DECLS
+
+#endif /* __GST_RTP_G722_DEPAY_H__ */
--- /dev/null
+/* GStreamer
+ * Copyright (C) <2010> Wim Taymans <wim.taymans@gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+
+#ifdef HAVE_CONFIG_H
+# include "config.h"
+#endif
+
+#include <string.h>
+
+#include <gst/audio/audio.h>
+#include <gst/audio/multichannel.h>
+#include <gst/rtp/gstrtpbuffer.h>
+
+#include "gstrtpg722pay.h"
+#include "gstrtpchannels.h"
+
+GST_DEBUG_CATEGORY_STATIC (rtpg722pay_debug);
+#define GST_CAT_DEFAULT (rtpg722pay_debug)
+
+static GstStaticPadTemplate gst_rtp_g722_pay_sink_template =
+GST_STATIC_PAD_TEMPLATE ("sink",
+ GST_PAD_SINK,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS ("audio/G722, " "rate = (int) 16000, " "channels = (int) 1")
+ );
+
+static GstStaticPadTemplate gst_rtp_g722_pay_src_template =
+GST_STATIC_PAD_TEMPLATE ("src",
+ GST_PAD_SRC,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS ("application/x-rtp, "
+ "media = (string) \"audio\", "
+ "encoding-name = (string) \"G722\", "
+ "payload = (int) " GST_RTP_PAYLOAD_G722_STRING ", "
+ "clock-rate = (int) 8000")
+ );
+
+static gboolean gst_rtp_g722_pay_setcaps (GstBaseRTPPayload * basepayload,
+ GstCaps * caps);
+static GstCaps *gst_rtp_g722_pay_getcaps (GstBaseRTPPayload * rtppayload,
+ GstPad * pad);
+
+GST_BOILERPLATE (GstRtpG722Pay, gst_rtp_g722_pay, GstBaseRTPAudioPayload,
+ GST_TYPE_BASE_RTP_AUDIO_PAYLOAD);
+
+static void
+gst_rtp_g722_pay_base_init (gpointer klass)
+{
+ GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
+
+ gst_element_class_add_pad_template (element_class,
+ gst_static_pad_template_get (&gst_rtp_g722_pay_src_template));
+ gst_element_class_add_pad_template (element_class,
+ gst_static_pad_template_get (&gst_rtp_g722_pay_sink_template));
+
+ gst_element_class_set_details_simple (element_class, "RTP audio payloader",
+ "Codec/Payloader/Network",
+ "Payload-encode Raw audio into RTP packets (RFC 3551)",
+ "Wim Taymans <wim.taymans@gmail.com>");
+}
+
+static void
+gst_rtp_g722_pay_class_init (GstRtpG722PayClass * klass)
+{
+ GstBaseRTPPayloadClass *gstbasertppayload_class;
+
+ gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
+
+ gstbasertppayload_class->set_caps = gst_rtp_g722_pay_setcaps;
+ gstbasertppayload_class->get_caps = gst_rtp_g722_pay_getcaps;
+
+ GST_DEBUG_CATEGORY_INIT (rtpg722pay_debug, "rtpg722pay", 0,
+ "G722 RTP Payloader");
+}
+
+static void
+gst_rtp_g722_pay_init (GstRtpG722Pay * rtpg722pay, GstRtpG722PayClass * klass)
+{
+ GstBaseRTPAudioPayload *basertpaudiopayload;
+
+ basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (rtpg722pay);
+
+ /* tell basertpaudiopayload that this is a sample based codec */
+ gst_base_rtp_audio_payload_set_sample_based (basertpaudiopayload);
+}
+
+static gboolean
+gst_rtp_g722_pay_setcaps (GstBaseRTPPayload * basepayload, GstCaps * caps)
+{
+ GstRtpG722Pay *rtpg722pay;
+ GstStructure *structure;
+ gint rate, channels, clock_rate;
+ gboolean res;
+ gchar *params;
+ GstAudioChannelPosition *pos;
+ const GstRTPChannelOrder *order;
+ GstBaseRTPAudioPayload *basertpaudiopayload;
+
+ basertpaudiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (basepayload);
+ rtpg722pay = GST_RTP_G722_PAY (basepayload);
+
+ structure = gst_caps_get_structure (caps, 0);
+
+ /* first parse input caps */
+ if (!gst_structure_get_int (structure, "rate", &rate))
+ goto no_rate;
+
+ if (!gst_structure_get_int (structure, "channels", &channels))
+ goto no_channels;
+
+ /* get the channel order */
+ pos = gst_audio_get_channel_positions (structure);
+ if (pos)
+ order = gst_rtp_channels_get_by_pos (channels, pos);
+ else
+ order = NULL;
+
+ if (rate == 16000)
+ clock_rate = 8000;
+
+ gst_basertppayload_set_options (basepayload, "audio", TRUE, "G722",
+ clock_rate);
+ params = g_strdup_printf ("%d", channels);
+
+ if (!order && channels > 2) {
+ GST_ELEMENT_WARNING (rtpg722pay, STREAM, DECODE,
+ (NULL), ("Unknown channel order for %d channels", channels));
+ }
+
+ if (order && order->name) {
+ res = gst_basertppayload_set_outcaps (basepayload,
+ "encoding-params", G_TYPE_STRING, params, "channels", G_TYPE_INT,
+ channels, "channel-order", G_TYPE_STRING, order->name, NULL);
+ } else {
+ res = gst_basertppayload_set_outcaps (basepayload,
+ "encoding-params", G_TYPE_STRING, params, "channels", G_TYPE_INT,
+ channels, NULL);
+ }
+
+ g_free (params);
+ g_free (pos);
+
+ rtpg722pay->rate = rate;
+ rtpg722pay->channels = channels;
+
+ /* octet-per-sample is 1 * channels for G722 */
+ gst_base_rtp_audio_payload_set_samplebits_options (basertpaudiopayload,
+ 4 * rtpg722pay->channels);
+
+ return res;
+
+ /* ERRORS */
+no_rate:
+ {
+ GST_DEBUG_OBJECT (rtpg722pay, "no rate given");
+ return FALSE;
+ }
+no_channels:
+ {
+ GST_DEBUG_OBJECT (rtpg722pay, "no channels given");
+ return FALSE;
+ }
+}
+
+static GstCaps *
+gst_rtp_g722_pay_getcaps (GstBaseRTPPayload * rtppayload, GstPad * pad)
+{
+ GstCaps *otherpadcaps;
+ GstCaps *caps;
+
+ otherpadcaps = gst_pad_get_allowed_caps (rtppayload->srcpad);
+ caps = gst_caps_copy (gst_pad_get_pad_template_caps (pad));
+
+ if (otherpadcaps) {
+ if (!gst_caps_is_empty (otherpadcaps)) {
+ GstStructure *structure;
+
+ structure = gst_caps_get_structure (otherpadcaps, 0);
+
+ gst_caps_set_simple (caps, "channels", G_TYPE_INT, 1, NULL);
+ gst_caps_set_simple (caps, "rate", G_TYPE_INT, 16000, NULL);
+ }
+ gst_caps_unref (otherpadcaps);
+ }
+ return caps;
+}
+
+gboolean
+gst_rtp_g722_pay_plugin_init (GstPlugin * plugin)
+{
+ return gst_element_register (plugin, "rtpg722pay",
+ GST_RANK_NONE, GST_TYPE_RTP_G722_PAY);
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) <2010> Wim Taymans <wim.taymans@gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+
+#ifndef __GST_RTP_G722_PAY_H__
+#define __GST_RTP_G722_PAY_H__
+
+#include <gst/gst.h>
+#include <gst/rtp/gstbasertpaudiopayload.h>
+
+G_BEGIN_DECLS
+
+#define GST_TYPE_RTP_G722_PAY \
+ (gst_rtp_g722_pay_get_type())
+#define GST_RTP_G722_PAY(obj) \
+ (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTP_G722_PAY,GstRtpG722Pay))
+#define GST_RTP_G722_PAY_CLASS(klass) \
+ (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTP_G722_PAY,GstRtpG722PayClass))
+#define GST_IS_RTP_G722_PAY(obj) \
+ (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_G722_PAY))
+#define GST_IS_RTP_G722_PAY_CLASS(klass) \
+ (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_G722_PAY))
+
+typedef struct _GstRtpG722Pay GstRtpG722Pay;
+typedef struct _GstRtpG722PayClass GstRtpG722PayClass;
+
+struct _GstRtpG722Pay
+{
+ GstBaseRTPAudioPayload payload;
+
+ gint rate;
+ gint channels;
+};
+
+struct _GstRtpG722PayClass
+{
+ GstBaseRTPAudioPayloadClass parent_class;
+};
+
+GType gst_rtp_g722_pay_get_type (void);
+
+gboolean gst_rtp_g722_pay_plugin_init (GstPlugin * plugin);
+
+G_END_DECLS
+
+#endif /* __GST_RTP_G722_PAY_H__ */
projects / platform / upstream / gstreamer.git / commitdiff